You are correct. Sorry I didn't see the engage_media_proxy() in your script.
Ross Beer-2 wrote: > > Hi, > > I used the 'engage_media_proxy();' in the main routing. This should create > a > dialog and then enable the media proxy if needed. Please correct me if I > have misunderstood this feature. > > Regards, > > Ross > > 2009/10/29 osiris123d <[email protected]> > >> >> In your config I don't see you calling the use_media_proxy() function >> anywhere. This is needed in order to proxy the media. >> >> Do you have the OpenSIPS mediaproxy module installed and have the >> parameters >> set up? >> Check this link out. >> http://www.opensips.org/Resources/DocsTutorials#toc12 >> >> You are going to need to use use_media_proxy() a couple of places in your >> config depending on what you want to accomplish. >> >> >> >> Ross Beer-2 wrote: >> > >> > Hi, >> > >> > I am using MediaProxy to help get over some one way audio issues, >> however >> > it >> > appears to be causing more problems than it is fixing. >> > >> > When I make a call between two registered phones there is no audio at >> all, >> > but when I call a gateway audio passes correctly. >> > >> > Looking at the logs it indicates that it has RTP & RTCP for one phone >> but >> > only RTP for the other: >> > >> > >> ------------------------------------------------------------------------------------------------------------------------------------------------------------- >> > *media-relay[11366]: debug: Received new SDP offer >> > media-relay[11366]: mediaproxy.mediacontrol.StreamListenerProtocol >> > starting >> > on 50060 >> > media-relay[11366]: mediaproxy.mediacontrol.StreamListenerProtocol >> > starting >> > on 50061 >> > media-relay[11366]: mediaproxy.mediacontrol.StreamListenerProtocol >> > starting >> > on 50062 >> > media-relay[11366]: mediaproxy.mediacontrol.StreamListenerProtocol >> > starting >> > on 50063 >> > media-relay[11366]: debug: Added new stream: (audio) >> > 192.168.2.200:5638(RTP: Unknown, RTCP: Unknown) <-> <SERVER IP >> > ADDRESS>:50060 <-> <SERVER IP >> > ADDRESS>:50062 <-> Unknown (RTP: Unknown, RTCP: Unknown) >> > media-relay[11366]: debug: created new session >> > NWZmMmMwMzAxNDM5YjdiYTAwMDYxYTViNTllMTczMWI.: >> > **10002*[email protected]*<10002*[email protected]> >> > * (b62884c7) --> **10001*[email protected]* <10001*[email protected]> >> > *media-relay[11366]: debug: Got traffic information for stream: (audio) >> > 192.168.2.200:5638 (RTP: Unknown, RTCP: Unknown) <-> <SERVER IP >> > ADDRESS>:50060 <-> <SERVER IP ADDRESS>:50062 <-> Unknown (RTP: <Clients >> > Router IP>:57096, RTCP: Unknown) >> > media-dispatcher[11369]: debug: Issuing "update" command to relay at >> > <SERVER >> > IP ADDRESS> >> > media-relay[11366]: debug: updating existing session >> > NWZmMmMwMzAxNDM5YjdiYTAwMDYxYTViNTllMTczMWI.: >> > **10002*[email protected]*<10002*[email protected]> >> > * (b62884c7) --> **10001*[email protected]* <10001*[email protected]> >> > *media-relay[11366]: debug: Received updated SDP answer >> > media-relay[11366]: debug: Got initial answer from callee for stream: >> > (audio) 192.168.2.200:5638 (RTP: Unknown, RTCP: Unknown) <-> <SERVER IP >> > ADDRESS>:50060 <-> <SERVER IP ADDRESS>:50062 <-> >> 192.168.2.10:40022(RTP: >> > <Clients Router IP>:57096, RTCP: Unknown) >> > media-relay[11366]: debug: Got traffic information for stream: (audio) >> > 192.168.2.200:5638 (RTP: Unknown, RTCP: <Clients Router IP>:55671) <-> >> > <SERVER IP ADDRESS>:50060 <-> <SERVER IP ADDRESS>:50062 <-> >> > 192.168.2.10:40022 (RTP: <Clients Router IP>:57096, RTCP: Unknown) >> > media-relay[11366]: debug: Got traffic information for stream: (audio) >> > 192.168.2.200:5638 (RTP: <Clients Router IP>:55670, RTCP: <Clients >> Router >> > IP>:55671) <-> <SERVER IP ADDRESS>:50060 <-> <SERVER IP ADDRESS>:50062 >> <-> >> > 192.168.2.10:40022 (RTP: <Clients Router IP>:57096, RTCP: Unknown) >> > media-dispatcher[11369]: debug: Issuing "remove" command to relay at >> > <SERVER >> > IP ADDRESS> >> > media-relay[11366]: debug: removing session >> > NWZmMmMwMzAxNDM5YjdiYTAwMDYxYTViNTllMTczMWI.: >> > **10002*[email protected]*<10002*[email protected]> >> > * (b62884c7) --> **10001*[email protected]* <10001*[email protected]> >> > *media-relay[11366]: (Port 50060 Closed) >> > media-relay[11366]: (Port 50061 Closed) >> > media-relay[11366]: (Port 50062 Closed) >> > media-relay[11366]: (Port 50063 Closed) >> > media-dispatcher[11369]: debug: Got statistics: {'from_tag': >> 'b62884c7', >> > 'dialog_id': '841:447573368', 'start_time': 1256818436.3299999, >> > 'timed_out': >> > False, 'call_id': 'NWZmMmMwMzAxNDM5YjdiYTAwMDYxYTViNTllMTczMWI.', >> > 'to_tag': >> > '7t0mkzlie0', 'streams': [{'status': 'closed', 'caller_codec': 'G711u', >> > 'post_dial_delay': 1.252835989, 'callee_codec': '1016', 'start_time': >> 0, >> > 'caller_bytes': 82000, 'callee_bytes': 83600, 'caller_packets': 410, >> > 'end_time': 8, 'callee_remote': '<Clients Router IP>:57096', >> > 'caller_remote': '<Clients Router IP>:55670', 'media_type': 'audio', >> > 'callee_local': '<SERVER IP ADDRESS>:50062', 'timeout_wait': 0, >> > 'caller_local': '<SERVER IP ADDRESS>:50060', 'callee_packets': 418}], >> > 'duration': 8, 'to_uri': >> > **'10001*[email protected]'*<'10001*[email protected]'> >> > *, 'from_uri': **'10002*[email protected]'* <'10002*[email protected]'>*, >> > 'callee_ua': 'snom370/7.3.26', 'caller_ua': 'X-Lite Beta release 4.0 >> Beta >> > 2 >> > stamp 55091'}* >> > >> ------------------------------------------------------------------------------------------------------------------------------------------------------------- >> > >> > I am using the following OpenSIP's config: >> > >> > >> > >> > # main routing logic >> > >> > route{ >> > >> > # initial sanity checks -- messages with >> > >> > # max_forwards==0, or excessively long requests >> > >> > if (!mf_process_maxfwd_header("10")) { >> > >> > sl_send_reply("483","Too Many Hops"); >> > >> > exit; >> > >> > }; >> > >> > if (msg:len >= 2048 ) { >> > >> > sl_send_reply("513", "Message too big"); >> > >> > exit; >> > >> > }; >> > >> > # !! Nathelper >> > >> > # Special handling for NATed clients; first, NAT test is >> > >> > # executed: it looks for via!=received and RFC1918 addresses >> > >> > # in Contact (may fail if line-folding is used); also, >> > >> > # the received test should, if completed, should check all >> > >> > # vias for rpesence of received >> > >> > if (nat_uac_test("31")) >> > >> > { >> > >> > # Allow RR-ed requests, as these may indicate that >> > >> > # a NAT-enabled proxy takes care of it; unless it is >> > >> > # a REGISTER >> > >> > xlog("Behind a NAT\n"); >> > >> > if (is_method("REGISTER")) >> > >> > { >> > >> > fix_nated_register(); >> > >> > } >> > >> > fix_nated_contact(); >> > >> > force_rport(); # Add rport parameter to topmost Via >> > >> > #setbflag(6); # Mark as NATed >> > >> > }; >> > >> > # we record-route all messages -- to make sure that >> > >> > # subsequent messages will go through our proxy; that's >> > >> > # particularly good if upstream and downstream entities >> > >> > # use different transport protocol >> > >> > if (!is_method("REGISTER")) >> > >> > record_route(); >> > >> > if(is_method("INVITE")) >> > >> > { >> > >> > fix_nated_sdp("1"); >> > >> > create_dialog(); >> > >> > fix_nated_sdp("8"); >> > >> > engage_media_proxy(); >> > >> > } >> > >> > # subsequent messages withing a dialog should take the >> > >> > # path determined by record-routing >> > >> > if (loose_route()) { >> > >> > # mark routing logic in request >> > >> > append_hf("P-hint: rr-enforced\r\n"); >> > >> > route(1); >> > >> > exit; >> > >> > }; >> > >> > if (!uri==myself) { >> > >> > # mark routing logic in request >> > >> > append_hf("P-hint: outbound\r\n"); >> > >> > route(1); >> > >> > exit; >> > >> > }; >> > >> > # if the request is for other domain use UsrLoc >> > >> > # (in case, it does not work, use the following command >> > >> > # with proper names and addresses in it) >> > >> > if (uri==myself) >> > >> > { >> > >> > if (is_method("REGISTER")) >> > >> > { >> > >> > # Uncomment this if you want to use digest authentication >> > >> > #if (!www_authorize("siphub.org", "subscriber")) { >> > >> > # www_challenge("siphub.org", "0"); >> > >> > # return; >> > >> > #}; >> > >> > save("location"); >> > >> > exit; >> > >> > }; >> > >> > # native SIP destinations are handled using our USRLOC DB >> > >> > if (!lookup("location")) >> > >> > { >> > >> > # Local Device Not Found Send To Gateway >> > >> > rewritehostport("<gateway>:5065"); >> > >> > } >> > >> > }; >> > >> > append_hf("P-hint: usrloc applied\r\n"); >> > >> > route(1); >> > >> > } >> > >> > route[1] >> > >> > { >> > >> > # !! Nathelper >> > >> > # if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && >> > !search("^Route:")) >> > >> > # { >> > >> > # sl_send_reply("479", "We don't forward to private IP addresses"); >> > >> > # exit; >> > >> > # }; >> > >> > # NAT processing of replies; apply to all transactions (for example, >> > >> > # re-INVITEs from public to private UA are hard to identify as >> > >> > # NATed at the moment of request processing); look at replies >> > >> > t_on_reply("1"); >> > >> > # send it out now; use stateful forwarding as it works reliably >> > >> > # even for UDP2TCP >> > >> > if (!t_relay()) { >> > >> > sl_reply_error(); >> > >> > }; >> > >> > } >> > >> > # !! Nathelper >> > >> > onreply_route[1] >> > >> > { >> > >> > if (nat_uac_test("31")) >> > >> > { >> > >> > # Allow RR-ed requests, as these may indicate that >> > >> > # a NAT-enabled proxy takes care of it; unless it is >> > >> > # a REGISTER >> > >> > xlog("Reply Behind a NAT"); >> > >> > fix_nated_contact(); >> > >> > force_rport(); # Add rport parameter to topmost Via >> > >> > #setbflag(6); # Mark as NATed >> > >> > }; >> > >> > } >> > >> > _______________________________________________ >> > Users mailing list >> > [email protected] >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > >> > >> >> -- >> View this message in context: >> http://n2.nabble.com/MediaProxy-No-Audio-Problems-tp3911881p3913596.html >> Sent from the OpenSIPS - Users mailing list archive at Nabble.com. >> >> _______________________________________________ >> Users mailing list >> [email protected] >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- View this message in context: http://n2.nabble.com/MediaProxy-No-Audio-Problems-tp3911881p3919261.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
